SIP is not getting registered from external LAN

Asterisk Asterisk Support
1

Hi,

I had setup a call center software named VICIDIAL and i have installed asterisk server too on opensuse, my question is that i am facing a problem with SIP registration. I am not able make SIP registration i have assigned a public IP Address to server that is 115.252.182.134 through IAX it is working and i"ll be able to make a calls but not getting on SIP i had add these configuration

nano /etc/asterisk/sip_nat.cof
externip=115…
localnet=192.168.12.0/255.255.255.0
nat=yes
qualify=yes
when i use sip debug the output was

(11 headers 0 lines) —
Really destroying SIP dialog ‘0ecbeb5c3f17c5ad59088b186f382041@115…’ Method: OPTIONS

<— SIP read from 115…:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 115…:5060;branch=z9hG4bK6ca0ebe6;received=115…;rport=5060
From: “asterisk” sip:asterisk@115.......;tag=as5cf1bc06
To: sip:testcarrier@115.......;cpd=on;tag=as5cf1bc06
Call-ID: 03d9108224101a42660fdc2f06b92494@115…
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: sip:asterisk@115.......
Accept: application/sdp
Content-Length: 0

<------------->
— (12 headers 0 lines) —
Really destroying SIP dialog ‘03d9108224101a42660fdc2f06b92494@115…’ Method: OPTIONS

— (12 headers 0 lines) —
Really destroying SIP dialog ‘5c8de29309c3ea4620e86e4e69910426@115…’ Method: OPTIONS

<— SIP read from 115…:4569 —>

<------------->

<— SIP read from 115…:4569 —>

<------------->
[Dec 24 12:13:54] WARNING[27426]: chan_sip.c:7695 determine_firstline_parts: Bad request protocol
== Parsing ‘/etc/asterisk/manager.conf’: Found
== Manager ‘sendcron’ logged on from 127.0.0.1
== Parsing ‘/etc/asterisk/manager.conf’: Found
== Manager ‘sendcron’ logged on from 127.0.0.1
== Manager ‘sendcron’ logged off from 127.0.0.1
== Manager ‘sendcron’ logged off from 127.0.0.1

<— SIP read from 115…:4569 —>

<------------->
[Dec 24 12:14:04] WARNING[27426]: chan_sip.c:7695 determine_firstline_parts: Bad request protocol
== Parsing ‘/etc/asterisk/manager.conf’: Found
== Manager ‘sendcron’ logged on from 127.0.0.1
== Manager ‘sendcron’ logged off from 127.0.0.1
Really destroying SIP dialog ‘6bd27e1d6d7b4e150bcb327106611e14@115…’ Method: REGISTER
Really destroying SIP dialog ‘33220f1543d301a5565c33ae08dd85c0@115…’ Method: REGISTER

<— SIP read from 115…:4569 —>

Actually on of my client wants make a call using SIP and we have developed a mobile based SIP application it should use our configuration but the problem is SIP is not getting registered what changes do i need change more . Please help on this I shall be very thankful to you people if you provide me best solution.

Thank You,
Shakir.

2

Your trace contains no register attempt.

Your trace does not contain the response.

The OPTIONS packet appears to be coming from the public address of the server that is receiving it - you seem to have a looping configuration.

What is supposed to be registering with what?

You appear to be receiving non-SIP packets on your SIP port.

/etc/asterisk/sip_nat.cof is not a standard Asterisk configuration file; it will be ignored unless you include code in a standard configuration file to invoke it.

Your configuration information contains no configuration relevant to registrations.

Why do you have nat=yes?

I hope this is an inbound centre!

3

Hi,

I had setup a call center software named VICIDIAL and i have installed asterisk server too on opensuse, my question is that i am facing a problem with SIP registration. I am not able make SIP registration i have assigned a public IP Address to server that is 115.252.182.134 through IAX it is working and i"ll be able to make a calls but not getting on SIP i had add these configuration

nano /etc/asterisk/sip_nat.cof
externip=115…
localnet=192.168.12.0/255.255.255.0
nat=yes
qualify=yes
when i use sip debug the output was

(11 headers 0 lines) —
Really destroying SIP dialog ‘0ecbeb5c3f17c5ad59088b186f382041@115…’ Method: OPTIONS

<— SIP read from 115…:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 115…:5060;branch=z9hG4bK6ca0ebe6;received=115…;rport=5060
From: “asterisk” sip:asterisk@115.......;tag=as5cf1bc06
To: sip:testcarrier@115.......;cpd=on;tag=as5cf1bc06
Call-ID: 03d9108224101a42660fdc2f06b92494@115…
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: sip:asterisk@115.......
Accept: application/sdp
Content-Length: 0

<------------->
— (12 headers 0 lines) —
Really destroying SIP dialog ‘03d9108224101a42660fdc2f06b92494@115…’ Method: OPTIONS

— (12 headers 0 lines) —
Really destroying SIP dialog ‘5c8de29309c3ea4620e86e4e69910426@115…’ Method: OPTIONS

<— SIP read from 115…:4569 —>

<------------->

<— SIP read from 115…:4569 —>

<------------->
[Dec 24 12:13:54] WARNING[27426]: chan_sip.c:7695 determine_firstline_parts: Bad request protocol
== Parsing ‘/etc/asterisk/manager.conf’: Found
== Manager ‘sendcron’ logged on from 127.0.0.1
== Parsing ‘/etc/asterisk/manager.conf’: Found
== Manager ‘sendcron’ logged on from 127.0.0.1
== Manager ‘sendcron’ logged off from 127.0.0.1
== Manager ‘sendcron’ logged off from 127.0.0.1

<— SIP read from 115…:4569 —>

<------------->
[Dec 24 12:14:04] WARNING[27426]: chan_sip.c:7695 determine_firstline_parts: Bad request protocol
== Parsing ‘/etc/asterisk/manager.conf’: Found
== Manager ‘sendcron’ logged on from 127.0.0.1
== Manager ‘sendcron’ logged off from 127.0.0.1
Really destroying SIP dialog ‘6bd27e1d6d7b4e150bcb327106611e14@115…’ Method: REGISTER
Really destroying SIP dialog ‘33220f1543d301a5565c33ae08dd85c0@115…’ Method: REGISTER

<— SIP read from 115…:4569 —>

when i used sip show settings the out is

Global Settings:

SIP Port: 5060
Bindaddress: 0.0.0.0
Videosupport: No
AutoCreatePeer: No
Allow unknown access: No
Allow subscriptions: Yes
Allow overlap dialing: No
Promsic. redir: No
SIP domain support: No
Call to non-local dom.: Yes
URI user is phone no: No
Our auth realm asterisk
Realm. auth: No
Always auth rejects: No
Call limit peers only: Yes
Direct RTP setup: No
User Agent: Asterisk PBX
MWI checking interval: 10 secs
Reg. context: (not set)
Caller ID: asterisk
From: Domain:
Record SIP history: Off
Call Events: On
IP ToS SIP: none
IP ToS RTP audio: none
IP ToS RTP video: none
T38 fax pt UDPTL: No
RFC2833 Compensation: No
SIP realtime: Disabled

Global Signalling Settings:

Codecs: 0x6 (gsm|ulaw)
Codec Order: ulaw:20,gsm:20
T1 minimum: 100
No premature media: No
Relax DTMF: Yes
Compact SIP headers: No
RTP Keepalive: 0 (Disabled)
RTP Timeout: 360
RTP Hold Timeout: 0 (Disabled)
MWI NOTIFY mime type: application/simple-message-summary
DNS SRV lookup: Yes
Pedantic SIP support: No
Reg. min duration 60 secs
Reg. max duration: 3600 secs
Reg. default duration: 120 secs
Outbound reg. timeout: 20 secs
Outbound reg. attempts: 0
Notify ringing state: Yes
Notify hold state: Yes
SIP Transfer mode: open
Max Call Bitrate: 384 kbps
Auto-Framing: No

Default Settings:

Context: trunkinbound
Nat: Always
DTMF: rfc2833
Qualify: 2000
Use ClientCode: No
Progress inband: No
Language: en
MOH Interpret: default
MOH Suggest: default
Voice Mail Extension: asterisk
Forward Detected Loops: Yes

please have look on it and do let me know if any changes has to be made according to you to get sip registered from external Network.

Actually on of my client wants make a call using SIP and we have developed a mobile based SIP application it should use our configuration but the problem is SIP is not getting registered what changes do i need change more . Please help on this I shall be very thankful to you people if you provide me best solution.